WO1996004653A1 - Procede et systeme de traitement de signaux permettant de remplacer des blocs ne pouvant pas etre corriges dans un recepteur, destines a des signaux audio codes en bloc - Google Patents

Procede et systeme de traitement de signaux permettant de remplacer des blocs ne pouvant pas etre corriges dans un recepteur, destines a des signaux audio codes en bloc Download PDF

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Publication number
WO1996004653A1
WO1996004653A1 PCT/DE1995/000997 DE9500997W WO9604653A1 WO 1996004653 A1 WO1996004653 A1 WO 1996004653A1 DE 9500997 W DE9500997 W DE 9500997W WO 9604653 A1 WO9604653 A1 WO 9604653A1
Authority
WO
WIPO (PCT)
Prior art keywords
signal
dsp
substitution
audio
processing arrangement
Prior art date
Application number
PCT/DE1995/000997
Other languages
German (de)
English (en)
Inventor
Dieter Leckschat
Original Assignee
Siemens Aktiengesellschaft
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Siemens Aktiengesellschaft filed Critical Siemens Aktiengesellschaft
Priority to JP50608896A priority Critical patent/JP3578461B2/ja
Priority to DE59510286T priority patent/DE59510286D1/de
Priority to EP95926833A priority patent/EP0775360B1/fr
Priority to AU31084/95A priority patent/AU694542B2/en
Priority to CA002196565A priority patent/CA2196565C/fr
Priority to US08/776,725 priority patent/US6134522A/en
Publication of WO1996004653A1 publication Critical patent/WO1996004653A1/fr

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Classifications

    • G—PHYSICS
    • G11—INFORMATION STORAGE
    • G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
    • G11B20/00—Signal processing not specific to the method of recording or reproducing; Circuits therefor
    • G11B20/10—Digital recording or reproducing
    • G11B20/18—Error detection or correction; Testing, e.g. of drop-outs
    • G11B20/1806—Pulse code modulation systems for audio signals

Definitions

  • transmitting and receiving devices are used for message processing and transmission.
  • the message generated by the message source is transmitted from the sending device to the receiving device via a message channel, which then sends the received message to the message sink.
  • the message processing and transmission can take place in a preferred transmission direction or in both transmission directions (duplex operation).
  • Message is a superordinate term that stands for both the meaning (information) and the physical representation (signal).
  • Signals can e.g. B. (1) pictures,
  • the type of transmission according to (1) ... (3) is usually characterized by continuous (analog) signals, while the type of transmission according to (4) usually gives rise to discontinuous signals (eg pulses, digital signals).
  • the present invention relates primarily to the
  • Either continuous signals occur as possible signal forms for an audio message system. or a mixture of continuous and discontinuous signals using A / D or D / A converters.
  • Message type-specific devices are required for the "Send” and "Receive" functions. The question of which of these devices will ultimately be used depends, among other things, on which message channel is used as the basis in the audio message system.
  • the present invention therefore relates primarily to telecommunication systems which have a wireless telecommunication channel.
  • Telecommunication systems with such a structure are, for example, cordless telephones according to the DECT standard (Digital European Cordless Telecommunication, - cf. (1) European Telecommunication Standard; prETS 300 175-1 ...
  • the DECT cordless telephone and the GSM mobile radio telephone are audio messaging systems in which audio signals coded in blocks - e.g. B. signals coded according to the TDMA or CDMA method (Time Division Multiple Access or Code Division Multiple Access).
  • the message transmitted with these telephones generally consists of a mixture of continuous and discontinuous signals. This signal mix is created through the use of analog / digital or. Digital / analog converters.
  • FIG. 1 shows a DECT cordless telephone with a cordless base station FT (Fixed Termination), to which a maximum of twelve cordless mobile parts PT1 ... PT12 (Portable Termination) 5 are assigned for cordless telecommunication via a radio channel FK.
  • FT Fixed Termination
  • Cordless base stations with such a structure are available under the product name "Gigaset 952" - cf. DE-Z: Funkschau 12/1993, pages 24 and 25; "Digital freedom - Gigaset 952: the first DECT telephone”; Author: G. Weckwerth - launched in 1993.
  • This structure is essentially also from DE-Z: Funkschau 10/1993; Pages 74 to 77; Title: “Communicate digitally with Philips DECT-DECT chipset”; Author: Dr. J. Nieder and WO 94/10812 ( Figure 1 with the associated description) previously known.
  • Figure 2 shows the basic structure of the DECT-specific cordless handset PT1 ... PT12, as it is used for the transmission of voice messages in the cordless telephone.
  • Cordless handsets with such a structure are also under the product name "Gigaset 952" - cf. DE-Z: radio
  • a first possibility known from WO 94/10769, consists in "muting" the corresponding signal block identified as faulty, ie. H. to change the code accordingly, for example by a sequence of zero values.
  • This method is used today with digital DECT cordless telephones, such as the Gigaset 952.
  • DSP Digital Signal Processor
  • the object on which the invention is based is to improve the transmission quality of audio signals coded in blocks in audio message systems when transmission errors occur in such a way that the need for computing power and thus the costs are minimal and there is no additional delay in the data to be transmitted Audio signal occurs.
  • the idea on which the invention is based is to replace pauses in the audio signal caused by transmission errors with a previously generated pause-specific substitution signal.
  • the substitution signal is generated by temporarily storing the signal section immediately preceding a current signal section of the audio signal and, in the event that the current signal section is disturbed, being inserted into the gap created by the disturbance.
  • This procedure lends itself simply because with audio signals (music or speech signals) there is a strong correlation between signal sections that are close in time. The reason for this is that the sound generation mechanisms (eg string vibration during music generation, movements in the vocal tract during speech generation etc.) have a certain mechanical inertia. If one compares successive signal sections of the audio signal in the order of magnitude of 10 to 20 ms, one almost always finds a very great similarity in the time signal (FIG. 3).
  • a low-pass filter is preferably used.
  • the low-pass filter has a smoothing effect in the time domain, while unnatural components of high frequency are attenuated in the frequency domain.
  • the associated impairment of the audio signal to be transmitted can be tolerated if the low-pass filter, which is preferably designed as a digital filter, does not cut the audio signal too strongly.
  • the tuning of the filter can therefore be seen as a compromise that is optimized from a psychoacoustic point of view. In addition, it must be demanded that the filter can be switched on and off without interference.
  • FIG. 4 the occurrence of discontinuities in the speech signal when replacing time periods (arrows),
  • FIG. 5 shows the modified topology of the cordless handset PT to improve the transmission quality of TDMA-specific (Time Division Multiple Access), DECT voice signals in DECT cordless phones when transmission errors occur,
  • TDMA-specific Time Division Multiple Access
  • DECT voice signals in DECT cordless phones when transmission errors occur
  • FIG. 6 shows the structure of a simple filter for suppressing substitution-related artifacts of the speech signal
  • FIG. 8 shows a DECT speech signal disturbed over three (10 ms) periods
  • FIG. 3 shows the time course of a speech signal SS a for the vowel "a", which is divided into several (10 ms) signal sections. With this temporal division of the speech signal SS a , the similarity of neighboring time segments becomes visible. This correlation between sections of the speech signal SS a which are close together is given by the fact that the sound generation mechanisms (movements in the vowel rakt) have a certain mechanical inertia.
  • FIG. 4 shows the same speech signal SS a for a different time axis, starting from the speech signal SS a for the voice "a" according to FIG. 3.
  • a time segment has been replaced in the speech signal according to FIG. 4 by copying the previous time segment. Because of this substitution, points of discontinuity have arisen at the points marked with the arrows, which can be heard as crack-like interferences during the electroacoustic signal conversion.
  • FIG. 5 shows, starting from FIG. 2, the modified topology of the cordless handset PT to improve the transmission quality of TDMA-specific (time division Multiple Access), DECT voice signals in DECT cordless phones when transmission errors occur.
  • TDMA-specific time division Multiple Access
  • DECT voice signals in DECT cordless phones when transmission errors occur.
  • the transmission errors often occur in border areas in the radio message transmission of the DECT cordless telephones, so that DECT-specific burst or information losses occur in these areas.
  • the modified cordless handset PT has a digital signal processor DSP, which on a transmission path US of the cordless handset PT - from an antenna ANT with a downstream radio part FKT (transmitter / receiver) to an earphone HK in the direction of reception or from a microphone MIK to the antenna ANT in the transmission direction - is arranged between a signal control device BMC (burst mode controller) and a signal conversion device SUE (codec, AD / DA converter).
  • the digital signal processor DSP is controlled by a function / sequence control device MIC (DECT microcontroller) that is specific to the mobile part.
  • the digital signal processor DSP can improve the DECT voice signals transmitted to the transmission path US, which is sometimes disturbed, the digital signal processor DSP has several program modules, (1) a first program module PM1 for generating and buffering a substitution signal corrected with the speech signal, (2) a second program module PM2 for determining at least one first incorrectly transmitted signal section in the speech signal,
  • the fourth program module PM4 will digits in the DECT voice signal are digitally filtered out.
  • the fourth program module PM4 is therefore nothing more than a digital filter in its preferred embodiment.
  • the jump-like discontinuities at the starting points of the temporal signal sections are smoothed out by the digital filter.
  • FIG. 6 shows the structure of a digital filter implemented by the PM4 program module according to FIG.
  • this digital filter is designed as a first order recursive filter (IIR filter; infinite impulse response filter).
  • the recursive filter has a filter function H ( ⁇ ) that
  • FIG. 7 shows the transfer function measured on a real low-pass filter of the first order.
  • the resulting band limitation at 4 kHz results from the bandwidth of the voice signal transmitted with the DECT-specific cordless handset PT at a scanning rate of 8 kHz. At the highest frequencies, this filter lowers the signal by almost 20 dB.
  • FIG. 8 shows a voice signal in which the DECT voice signal is disturbed over three (10 ms) periods (time range on the time axis between 4425 ms and 4455 ms).
  • the individual (10th ms) period corresponds to the duration of a TDMA time division multiplex frame in DECT cordless signal transmission.
  • the modified voice signal shown in FIG. 9 ultimately results at the output of the signal conversion device SUE according to FIG.
  • the difference resulting from the original speech signal according to FIG. 8 is the sole result of the processing of the original speech signal in the digital signal processor DSP according to FIGS. 5 and 6. That in the digital signal processor DSP due to the program modules PM1 ... PM4 functional steps are:
  • the digital signal processor DSP only needs the last buffer value stored.
  • the two coefficients a ⁇ _, bg then only need to be converted. If several signal sections are disturbed in the speech signal, as in the speech signal according to FIG. 8, this error is eliminated by correspondingly inserting the last one transmitted without interference.
  • this method can only be used to a limited extent - for (10 ms) signal sections, this limit is approximately (empirical values) with a duration of 50 ms. Using the method makes no sense if error-free speech signals are no longer received. A constant repetition of the last disturbed signal section would lead to an unnatural auditory impression.
  • the described method is modified in such a way that, after some disturbed signal sections have been replaced by the last signal section transmitted without error, the speech signal subsequently has a time constant of z. B. 20 ms is hidden.
  • This operation can be carried out by the digital signal processor DSP without great computation effort.
  • the digital filter to vary in time. This can be done, for example, by lowering the cutoff frequency of the filter and thus increasing the effect of the filter. Due to the properties of the digital signal processor DSP described above, it can differentiate how many signal sections (DECT bursts) have been transmitted incorrectly and accordingly react differently depending on the duration of the disturbed signal section.

Landscapes

  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Signal Processing (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
  • Mobile Radio Communication Systems (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)

Abstract

L'invention vise à améliorer la qualité de transmission de signaux codés en blocs dans des systèmes de communications audio, par ex. dans des téléphones sans fil DECT, en cas d'apparition d'erreurs de transmission, de façon à réduire à un minimum la puissance de calcul nécessaire et par conséquent les coûts correspondants, et à éviter tout retard supplémentaire du signal audio à transmettre. A cet effet, les interruptions dans le signal audio dues à des erreurs de transmission sont remplacées par un signal de substitution généré au préalable et spécifique à l'interruption concernée. Le signal de substitution est généré, dans le plus simple des cas, de la manière suivante: le segment de signal précédant immédiatement le segment de signal actuel est mis en mémoire tampon, puis au cas où le segment de signal actuel est perturbé, intégré dans l'espace vide produit par la perturbation.
PCT/DE1995/000997 1994-08-02 1995-08-01 Procede et systeme de traitement de signaux permettant de remplacer des blocs ne pouvant pas etre corriges dans un recepteur, destines a des signaux audio codes en bloc WO1996004653A1 (fr)

Priority Applications (6)

Application Number Priority Date Filing Date Title
JP50608896A JP3578461B2 (ja) 1994-08-02 1995-08-01 音声通信システムにおけるブロック符号化音声信号の処理方法及び装置
DE59510286T DE59510286D1 (de) 1994-08-02 1995-08-01 Signalverarbeitungsverfahren und -anordnung zum ersetzen nichtkorrigierbarer blöcke in einem empfänger für blockweise kodierte audiosignale
EP95926833A EP0775360B1 (fr) 1994-08-02 1995-08-01 Procede et systeme de traitement de signaux permettant de remplacer des blocs ne pouvant pas etre corriges dans un recepteur, destines a des signaux audio codes en bloc
AU31084/95A AU694542B2 (en) 1994-08-02 1995-08-01 Signal-processing method and device for substituting non-correctable blocks in a receiver for block-coded audio signals
CA002196565A CA2196565C (fr) 1994-08-02 1995-08-01 Procede et systeme de traitement de signaux permettant de remplacer des blocs ne pouvant pas etre corriges dans un recepteur, destines a des signaux audio codes en bloc
US08/776,725 US6134522A (en) 1994-08-02 1995-08-01 Signal processing method and arrangement for substitution or erroneous signals in a block-coded audio signals of an audio communication system

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
DE4427351A DE4427351C1 (de) 1994-08-02 1994-08-02 Signalverarbeitungsverfahren und -anordnung für blockweise codierte Sprachsignale eines Nachrichtensystems
DEP4427351.7 1994-08-02

Publications (1)

Publication Number Publication Date
WO1996004653A1 true WO1996004653A1 (fr) 1996-02-15

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PCT/DE1995/000997 WO1996004653A1 (fr) 1994-08-02 1995-08-01 Procede et systeme de traitement de signaux permettant de remplacer des blocs ne pouvant pas etre corriges dans un recepteur, destines a des signaux audio codes en bloc

Country Status (9)

Country Link
US (1) US6134522A (fr)
EP (1) EP0775360B1 (fr)
JP (1) JP3578461B2 (fr)
CN (1) CN1149568C (fr)
AU (1) AU694542B2 (fr)
CA (1) CA2196565C (fr)
DE (2) DE4427351C1 (fr)
ES (1) ES2179878T3 (fr)
WO (1) WO1996004653A1 (fr)

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US6853867B1 (en) 1998-12-30 2005-02-08 Schneider Automation Inc. Interface to a programmable logic controller
US6651207B1 (en) * 1999-08-20 2003-11-18 Siemens Information & Communication Mobile, Llc Method and system for improving voice quality in cordless communications
US6278742B1 (en) * 1999-11-19 2001-08-21 Siemens Information And Communication Mobile Llc. Method and system for power-conserving interference avoidance in communication between a mobile unit and a base unit in a wireless telecommunication system
US7028204B2 (en) * 2000-09-06 2006-04-11 Schneider Automation Inc. Method and apparatus for ethernet prioritized device clock synchronization
US20020167967A1 (en) * 2000-09-06 2002-11-14 Schneider Electric Method for managing bandwidth on an ethernet network
DE10130233A1 (de) * 2001-06-22 2003-01-02 Bosch Gmbh Robert Verfahren zur Störverdeckung bei digitaler Audiosignalübertragung
DE10139403A1 (de) * 2001-08-10 2003-02-27 Tenovis Gmbh & Co Kg Verfahren zur Überbrückung einer tonlosen Sprachpause während einer Telekommunikationsverbindung sowie Telekommunikationsanlage
US7142250B1 (en) * 2003-04-05 2006-11-28 Apple Computer, Inc. Method and apparatus for synchronizing audio and video streams
US20090037935A1 (en) * 2006-08-07 2009-02-05 Shoumen Saha Updating The Configuration of Container Documents
US8185830B2 (en) * 2006-08-07 2012-05-22 Google Inc. Configuring a content document for users and user groups
US8862254B2 (en) 2011-01-13 2014-10-14 Apple Inc. Background audio processing
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Also Published As

Publication number Publication date
CA2196565C (fr) 2002-06-25
US6134522A (en) 2000-10-17
AU3108495A (en) 1996-03-04
CN1157665A (zh) 1997-08-20
AU694542B2 (en) 1998-07-23
DE4427351C1 (de) 1996-03-28
DE59510286D1 (de) 2002-08-22
JPH09509545A (ja) 1997-09-22
CA2196565A1 (fr) 1996-02-15
EP0775360A1 (fr) 1997-05-28
CN1149568C (zh) 2004-05-12
EP0775360B1 (fr) 2002-07-17
JP3578461B2 (ja) 2004-10-20
ES2179878T3 (es) 2003-02-01

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